A video conference uses a set of interactive telecommunication technologies which allow two or more locations to interact via multi-way video and audio transmissions simultaneously. A video conference bridge exchanges video streams with clients over wide area network (WAN) links. The core technology used in a videoconferencing system is digital compression of audio and video streams in real time. The hardware or software that performs compression is called a codec (coder/decoder). Compression rates of up to 1:500 can be achieved.
Presentation and transmission of video and multimedia data, even when compressed using state of the art codecs, requires a large amount of bandwidth. For example, a 1280×720 30-frame/sec video stream can easily consume the bandwidth of an access link such as a 1.544 Megabit/second T1 link. Bandwidth on wide area network (WAN) links, especially at the service levels required for real-time communication, is expensive. When a video conference bridge processes a request to start a new conference or join an additional client to an existing conference, one of the factors that the bridge must consider is whether the additional video and audio streams will overload any WAN links. A simple test used is to determine whether a maximum bit rate of a new stream plus the sum of maximum bit rates of established streams is less than the link bit rate. However, since the actual bit rate of a video stream varies with time, this approach is inefficient as it can leave a fair amount of bandwidth unused. For example, while it may first appear that the addition of the new stream will overload the link, conditions may exist where the current stream requirements will lessen, e.g., muting audio, less movement in the room, etc. The result is that a new session may be denied when in fact unused bandwidth is or may become available. Therefore, what is needed is a more efficient way of allocating bandwidth among communication sessions such as video conferences.